Current service providing network systems, such as the IP Multimedia Subsystem as developed by the Third Generation Partnership Project (3GPP), are designed to provide IP Multimedia over mobile communication networks (3GPP TS 22.228, TS 23.218, TS 23.228, TS 24.228, TS 24.229, TS 29.228, TS 29.229, TS 29.328 and TS 29.329). For fixed broadband services, such as Voice over IP (VoIP), the ETSI TISPAN working group is further developing IMS (TS 29.229: IP Multimedia Call Control Protocol based on SIP and SDP).
Within the IMS architecture the basic end-user subscription functions and the IP session management are decoupled from the specific VoIP service functions, e.g. number analysis, CLIP/R, Call Waiting, Call Barring, Call Waiting, etc. These services are handled within one or more application servers which reside in the network.
IMS, which makes use of the Session Initiation Protocol (SIP) to set up and control client-to-client call services and client-to-server call services, provides the delivery of reliable VoIP services which meet the requirements regarding Quality of Services (QoS) and the regulatory demands for routing, privacy, security and legal interception.
Although IMS enables a large amount of multimedia services, it also has disadvantages, especially within the context of VoIP. Within the IMS standard, VoIP is only described in combination with the use of application servers in the network to maintain call state and offer the voice service features. The application servers hosting the VoIP services require user-specific configuration data for each service. Conventional IMS application servers are required to be involved in the call session for the complete duration of the call to maintain call state and will cause a substantial amount of SIP messaging to be exchanged between the various clients and servers involved. The capacity of an application server is dependent on the number of subscribers to the services, while the capacity of the IMS system scales with the volume of data traffic. Hence, the dimensioning of the network resources in such conventional IMS system is complex and based on forecasts of the VoIP service behaviour of the users is needed.
One way of solving these problems is an IMS-based call handling architecture wherein a predetermined number of the VoIP call services, which normally reside in the network, are located in the user terminals connected to the IMS. Such architecture, which hereafter will be referred to as the “flat” IMS architecture, is described in European patent application no. 080053937, which is hereby incorporated by reference into this application.
Within the flat IMS architecture, the initial Filter Criteria (iFC) in the user service profile of the Home Subscriber Server (HSS) may comprise information which determines whether or not a SIP message should be routed to a service located in a particular application server. The iFC may be defined according to the standard in paragraph B.2.2 of document TS 129 228, which is hereby incorporated by reference in this application. An iFC may comprise a Trigger Point, i.e. a Boolean flag determined by a set of conditions and the SIP URI of an application server the SIP request should be routed to in case of a received SIP message fulfils the condition(s) set by the Trigger point (Trigger point is TRUE). In case the Trigger point is FALSE, the SIP message will not be routed to the application server comprising the service identified in the iFC.
Hence, the iFC may be defined in such a way that a user terminal may be prevented from registering to services located in the application servers of the IMS which correspond to the call services present in the user terminal. This may be realized by setting the Trigger Point in the iFCs associated with the VoIP call services corresponding to the call services in the UE to FALSE. Hence, after registration, the iFCs in the service profile may determine the S-CSCF to route SIP messages via one or more application servers only when these application servers host services which are not present or active in the user terminal.
FIG. 1 depicts an exemplary flow diagram 100 of a VoIP call session in a flat IMS architecture, which includes the activation of a Call Forwarding service.
The core of the IMS may be formed by the conventional Call/Session Control Functions (CSCF) comprising amongst others a Proxy-CSCF (P-CSCF), an Interrogating-CSCF (I-CSCF) and a Serving-CSCF (S-CSCF). A first user terminal UE-A 102, a second user terminal UE-B 104 and a third user terminal UE-C 106 may comprise a predetermined number of originating VoIP services (OS) and/or terminating VoIP services (TS) (108,110,112) connected to a SIP client. Each user terminal may be registered with the flat IMS. Upon registration the user profile associated with each user terminal may instruct its serving S-CSCF not to route SIP messages via application servers for all services or at least one or more services which are already present in the user terminal. For each of these services the Trigger Point in its associated iFC may be set to FALSE.
The call session depicted in FIG. 1 starts with the first terminal UE-A 102 receiving a request for a call. Such request may be initiated e.g. by the user dialing a local number 3434343 of the second user terminal UE-B. The request triggers a service 108 residing in UE-A, e.g. a VoIP number normalization service. This service generates a normalized number of the user terminal UE-B, which is subsequently inserted as a R-URI in the header of the SIP INVITE message 114 generated by the SIP client of UE-A. The SIP message is then routed via a Session border Controller (SBC) and the P-CSCF to the S-CSCF 116 serving the first user terminal UE-A.
On the basis of the user profile retrieved from a Home Subscriber Server (HSS) 118 the S-CSCF may directly forward the SIP message to the I-CSCF of user terminal UE-B using ENUM and DNS. No routing to an VoIP application server takes place. Similarly, the S-CSCF 120 of the second user terminal UE-B 104 may directly route the SIP INVITE message to the SIP client of user terminal UE-B without addressing the one or more application servers connected to the IMS comprising the call services in the second user terminal UE-B. In response to the reception of the SIP message, a Call Forward service 110 located in the user terminal UE-B may be executed. The SIP client of UE-B thereafter sends the SIP INVITE message with the R-URI identifying the third user terminal UE-C (in this case a telephone number +31201234567) to the S-CSCF 122 serving UE-B.
In response, the S-CSCF 122 serving UE-B may identify the I-CSCF of the third user terminal UE-C using ENUM and DNS and directly forwards the SIP INVITE message via the I-CSCF to the S-CSCF serving UE-C 124, which subsequently forwards the SIP INVITE message on the basis of the user profile of UE-C to the SIP client of UE-C 106.
User terminal UE-C may confirm the establishment of a call session between UE-C and UE-B by sending a SIP response message, typically a SIP 200 OK message, back to UE-B and UE-B may confirm the establishment of the call session between UE-A and UE-B by sending a SIP 200 OK message back to UE-A (not shown in FIG. 1). In this way, a call session between UE-A and UE-C is established wherein the voice data are communicated over the connection using e.g. the RTP protocol.
Hence, service provisioning in the flat IMS architecture results in a significant signaling load reduction in the network, especially with regard to the IMS core and the application servers. It thus allows a very low cost solution of VoIP services. Within the flat IMS architecture however certain problems regarding service control may still be present.
In a conventional IMS the SIP client of UE-A inserts the identity of user terminal A in the FROM field of the SIP message header. The receiving SIP client of UE-B may present this identity as a Calling Line Identity to the called user of UE-B. The called user may trust this identity as being controlled by the logic of a VoIP application server of the serving telecom operator.
Furthermore, in a conventional IMS it is the service logic of a VoIP application server in the network that performs the forwarding actions to a further user terminal UE-C. In that case the VoIP service assumes separate call legs. For a call being forwarded, the first call leg is defined by the call between UE-A and UE-B and the second call leg is defined by the call between UE-B and UE-C, wherein the user of UE-A typically pays for the first call leg and the user of UE-B pays for the second call leg. Hence in a conventional IMS it is the service logic of a (VoIP) application server in the network that maintains the call state of the call between UE-A and UE-B and, between UE-B and UE-C.
In an IMS architecture wherein part of the VoIP services reside in the end-terminals however the identity of user terminal UE-A cannot be guaranteed as the content of the FROM field will be passed on transparently. As a result, the identity of UE-A may be changed by mistake or manipulated for fraud reasons and the user terminal UE-B can be mislead.
Moreover, forwarding actions will be performed by the SIP client EU-B located in the user terminal B. Hence, UE-B may reject an incoming call with a SIP response message, e.g. a SIP 302 Moved Temporarily message. This is illustrated by the flow diagram 200 of FIG. 2. Similar to the flow diagram in FIG. 1, a SIP INVITE 214 is sent by the SIP agent of user terminal UE-A 202 to the SIP agent of user terminal UE-B 206. In response to the SIP INVITE UE-B sends a SIP 302 Moved Temporarily message 216 back to UE-A. The SIP 302 response message instructs the UE-A to set-up a direct call session 218 between UE-A and UE-C. This redirection is out of the control of the user of UE-A and may lead to the situation that the user of UE-A is billed for the call to the forwarded call to user terminal UE-C. This may lead to unwanted payments, especially when the user terminal UE-C is a highly priced 900-number. Moreover, user terminal UE-B may return an error response code whereby user terminal UE-A becomes uncontrollable for its user.